In a remote presentation system developed based on a video conference system of the H.323 protocol put forward by the ITU-T, a new interaction flow is introduced to complete the interaction and negotiation of new characteristic parameters of remote presentation (such as a media collecting attribute, an encoding group attribute, a scenario collecting attribute and a simultaneous capability set attribute) and multi-code stream characteristic information (a code stream multiplexing mode and multiplexing information). Therefore, how to keep a relative independence from the existing H.323 protocol stack at a protocol signaling aspect, and enable the implementation of the new characteristics to have a good expandability, and be easier to implement an intercommunication with the remote presentation system based on a Session Initiation Protocol (SIP) proposed by the Internet Engineering Task Force (IETF) is an important problem required to be considered.
From a technical perspective, it can modify the existing H.323 protocol stack, expand parts of interaction flows and protocol parameters to complete the interaction and negotiation of new characteristic parameters of the remote presentation. For example, part of product implementations are a remote presentation system integrated from multiple H.323 video conference terminals, each terminal respectively performs calling and capability set interaction; or the interaction of parts of remote presentation ability parameters is implemented by extending an H.245 capability set. Since the characteristic parameters of the remote presentation and multi-code stream attributes are more complicated, a great modification to the original H.323 protocol will be caused only by means of expanding the H.323 protocol, it is extremely difficult to describe complicated parameters due to the limitation of the limitation of the H.323 protocol message structure,. In the scheme of modifying the existing H.323 protocol stack, it needs the manufacturers to change the protocol stack, which is very difficult from a business perspective. On the other hand, in order to be convenient for intercommunicating with the remote presentation system based on the SIP defined by the standards organization Internet Engineering Task Force (IETF), separately processing the newly added remote presentation new functions and the existing basic call session and capability negotiation is a better solution.
At present, there is also one implementation way which is based on H.323 protocol stack. and introduces an enterprise standard Telepresence Interoperability Protocol (TIP) put forward by Cisco The TIP defines a plurality of message fields to transmit the remote presentation characteristic parameters and identify the multi-path code stream information by using rules of a Real-time Transport Protocol (RTP) Control Protocol (RTCP), since there have been an RTCP channel and message transmission after a media channel is established in the original H.323 or SIP system, the introduction of the TIP protocol is only required to encapsulate and parse the newly defined parameters with the RTCP message, which relates to the interaction between the original H.323 protocol stack and the TIP protocol; but, the TIP protocol itself is an extension to the RTCP message, only can transmit a very simple kind of information such as a relation between the agent position and the code stream, and the more complicated remote presentation ability parameters are not considered, which has a very great limitation, and cannot solve the problem of interaction of more complicated new characteristics in the remote presentation system.
At present the ControLling mUltiple streams for tElepresence (CLUE) working group of the IETF are discussing new characteristic parameters of the remote presentation and multi-code stream attributes, and the traditional video conference product manufacturers need to consider how to introduce the contents of the CLUE protocol in the H.323 system.